CUCM and Avaya CS1000 SIP connection

                   Hello - looking for some help on a SIP trunk configuration between the 2 devices. Currently we are running CUCM 9.1 with Avaya Session Manager 6.3. We are having issues with the call completing from the CS 1000 to the CUCM. Below are the session traces from both the Session Manager and CUCM. The CS1000 currently has 4 digit extensions and the CUCM has 7 digit extensions. We translate the number in the Session Manager to send the 7 digits. If you could lead in the right direction I would appreciate it. WOuld this have something to do with the context coming out of Session Manager? It looks like CDP.UDP and then only the 4 digits and the CUCM needs the 7. I also attached the configurations guide used for this.
Session Manager trace:
   mil-ss-01                 CUCM 9
                       SM100                10.101.2.75             10.174.2.75
13:04:53.730 |<--OPTIONS-|           |           |           |           | (1) sip:10.5.1.30
13:04:53.731 |--200 OK-->|           |           |           |           | (1) 200 OK (OPTIONS)
13:04:53,734 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:53,734 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:56.183 |--INVITE-->|           |           |           |           | (2) T:7317;phone-context=cdp.udp F:anonymous@anonymous U:7317;phone-context=cdp.udp
13:04:56.184 |<--Trying--|           |           |           |           | (2) 100 Trying
13:04:56,185 |                  Remote host is trusted                   | Trusted
13:04:56,185 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:56,186 |                Applied ingress Adaptation                 | P-Asserted-Identity=<sip:[email protected]>, Request-URI=sip:[email protected], History-Info=<sip:[email protected]>;index=1, <sip:[email protected]>;index=1.1
13:04:56,186 |                Originating Location found                 | Location: mil-cs1000m-01
13:04:56,186 |        Try routing to determine if emergency call         | Location: mil-cs1000m-01
13:04:56,186 |                Request Dial Pattern route                 | for: sip:[email protected] Location: mil-cs1000m-01
13:04:56,186 |               Dial Pattern route parameters               | URI Domain: company.com Location: mil-cs1000m-01
13:04:56,186 |                 Trying Dial Pattern route                 | Domain: company.com Location: mil-cs1000m-01
13:04:56,186 |                    Dial Pattern found                     | for: 7317 Pattern: 7317
13:04:56,186 |                    Route Policy found                     | Pattern: 7317 RoutePolicyList: to_CUCM9
13:04:56,187 |                        Route found                        | for: sip:[email protected] SIPEntity: CUCM 9
13:04:56,187 |                     Entity Link found                     | SIPEntity: CUCM 9 EntityLink: mil-sessionmgr-01->TCP, biDirId=null, deny=false:5060
13:04:56,187 |                    Request Adaptation                     | Adapter: CUCM 9
13:04:56,188 |                 Applied egress Adaptation                 | NoAdaptationModuleExists=true, Request-URI=sip:[email protected];routeinfo=0-0, Remote-Party-ID=<sip:[email protected]>;party=calling;screen=no;privacy=off,
13:04:56,188 |                    Routing SIP request                    | SipEntity: CUCM 9 EntityLink: mil-sessionmgr-01->TCP:5060
13:04:56,189 |              No hostname resolution required              | Routing to: sip:10.5.131.12;transport=tcp;lr;phase=terminating
13:04:56.191 |           |--INVITE-->|           |           |           | (2) T:7317;phone-context=cdp.udp F:anonymous@anonymous U:7657317 P:terminating
13:04:56.196 |           |<--Trying--|           |           |           | (2) 100 Trying
13:04:56.198 |           |<--Not Fou-|           |           |           | (2) 404 Not Found
13:04:56.199 |           |----ACK--->|           |           |           | (2) sip:[email protected]
13:04:56,200 |                    Request Adaptation                     | Adapter: CUCM 9
13:04:56,201 |                    Request Adaptation                     | Adapter: CUCM 9
13:04:56,201 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:56.202 |<--Not Fou-|           |           |           |           | (2) 404 Not Found
13:04:56.203 |----ACK--->|           |           |           |           | (2) sip:7317
13:05:07,597 |                Remote host is not trusted                 | Host not trusted
13:05:07,597 |                Originating Location found                 | Location: mil-cs1000m-01
13:05:12.657 |           |<--------OPTIONS-------|           |           | (3) sip:10.5.2.51
13:05:12,659 |                Remote host is not trusted                 | Host not trusted
13:05:12,659 |                Originating Location found                 | Location: mil-cs1000m-01
13:05:12.660 |           |--------200 OK-------->|           |           | (3) 200 OK (OPTIONS)
13:05:16.877 |           |<--------------OPTIONS-------------|           | (4) sip:10.5.2.51
13:05:16,879 |                  Remote host is trusted                   | Trusted
13:05:16,879 |                    Request Adaptation                     | Adapter: mil-ss-01
13:05:16,879 |                Applied ingress Adaptation                 | P-Asserted-Identity=<sip:[email protected]>
13:05:16,879 |                Originating Location found                 | Location: sas-cs1000e-01
13:05:16.880 |           |--------------200 OK-------------->|           | (4) 200 OK (OPTIONS)
13:05:24.463 |           |<--------------------OPTIONS-------------------| (5) sip:10.5.2.51
13:05:24,465 |                Remote host is not trusted                 | Host not trusted
13:05:24,465 |                Originating Location found                 | Location: mil-cs1000m-01
13:05:24.466 |           |--------------------200 OK-------------------->| (5) 200 OK (OPTIONS)
CUCM Trace Invite:
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 3188 bytes:
[1179,NET]
INVITE sip:[email protected] SIP/2.0
P-AV-Message-Id: 1_1
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
History-Info: <sip:[email protected]>;index=1, <sip:[email protected]>;index=1.1
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=no;privacy=off
Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, PRACK, OPTIONS, INFO, SUBSCRIBE, UPDATE
Contact: <sip:00000000;[email protected]:5060;maddr=10.5.1.30;transport=tcp;user=phone;gsid=68cac530-5d21-11e3-8b45-78e3b505dc88>
Alert-Info: <cid:[email protected]>
Supported: 100rel, x-nortel-sipvc, replaces
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
Via: SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140
Via: SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
Record-Route: <sip:[email protected];transport=tcp;lr>
Record-Route: <sip:10.5.2.50:15060;transport=tcp;ibmsid=local.1372169047609_2400497_2400521;lr>
Record-Route: <sip:[email protected];transport=tcp;lr>
P-Charging-Vector: icid-value="68cac530-5d21-11e3-8b45-78e3b505dc88"
User-Agent: Nortel CS1000 SIP GW release_7.0 version_linux-6.50.00 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: <sip:[email protected]>
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
Max-Forwards: 66
CSeq: 1 INVITE
Content-Type: multipart/mixed;boundary=unique-boundary-1
Content-Length: 1063
Av-Global-Session-ID: 68cac530-5d21-11e3-8b45-78e3b505dc88
P-Location: SM;origlocname="mil-cs1000m-01";origsiglocname="mil-cs1000m-01";origmedialocname="mil-cs1000m-01";termlocname="Cisco BE6K";termsiglocname="Cisco BE6K";smaccounting="true"
--unique-boundary-1
Content-Type: application/sdp
SDP Message
====================================================
v=0
o=- 746 1 IN IP4 10.5.1.30
s=-
c=IN IP4 10.5.1.36
t=0 0
m=audio 5234 RTP/AVP 18 0 8 101 111
c=IN IP4 10.5.1.36
a=tcap:1 RTP/SAVP
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=pcfg:1 t=1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=ptime:20
a=sendrecv
--unique-boundary-1
Content-Type: application/x-nt-mcdn-frag-hex;version=linux-6.50.00;base=x2611
Content-Disposition: signal;handling=optional
0500bc05
0107130081900000a200
09090f00e9a4830001004000
1315070011fa0f00a10d02010102020100cc040000c56000
1e0403008183
4a1c0100180001001a011404000067353505000004000000000048710000
--unique-boundary-1
Content-Type: application/x-nt-epid-frag-hex;version=linux-6.50.00;base=x2611
Content-Disposition: signal;handling=optional
011201
3c:4a:92:f4:84:f4
--unique-boundary-1--
CUCM Trying Message:
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1180,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Content-Length: 0
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1181,NET]
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 623 bytes:
[1182,NET]
ACK sip:[email protected] SIP/2.0
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1180,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Content-Length: 0
CUCM not found message:
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1181,NET]
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
CUCM ACK message:
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 623 bytes:
[1182,NET]
ACK sip:[email protected] SIP/2.0
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0
Thanks.

This document worked for us between CUCM BE6000 ver 9.0 and the Avaya.
The main focus on the Cisco side is this: Page 37 - 41
5.4. Define SIP Trunk Security Profile
Expand System  Security Profile and select SIP Trunk Security Profile. Click to
configure a SIP Trunk Security Profile.
Enter the following values and use defaults for remaining fields:
 Name Enter name
 Description Enter a brief description
 Incoming Transport Type Verify “TCP+UDP” is selected
 Outgoing Transport Type Verify “TCP” is selected
 Accept Out-of-Dialog REFER Enter
 Accept Unsolicited Notification Enter
 Accept Replaces Header Enter
Click . The screen below shows SIP Trunk Security Profile for the sample configuration
5.5. Define SIP Profile
Expand Device  Device Settings and select SIP Profile. Click to configure a SIP
Profile.
Under SIP Profile Information section, enter the following values and use defaults for
remaining fields:
 Name Enter name
 Description Enter a brief description
 Default MTP Telephony Event Payload Type Enter “120”
 Disable Early Media on 180 Enter
Note: Disabling Early Media allows local ringback to be used.
Under Parameters used in Phone section, scroll to end of section and enter the following values
and use defaults for remaining fields:
 RFC 2543 Hold Enter
Click . The screen below shows SIP Profile for the sample configuration.

Similar Messages

How to create multiple sip trunks between cucm and cisco unified sip proxy

Dear Expert,
Is there a way to create multiple sip trunks between CUCM and Cisco Unified SIP Proxy (CUSP)? How to achieve it without creating multiple IP interfaces on the CUSP module.
CUCM: 8.5.1.10000-9
CUSP: 8.5.2
Thank you,
.wan

Hello Michael,
This SIP trunk is part of UCCE solution, which used between CVP, CUSP, and CUCM.
The requirements:
1) To have different codecs for different type of calls, as the phones are at few countries
2) To pass different number of digits from CUSP to CUCM for different call treatments
.wan

SIMPLE QUESTION: LYNC 2010 -- AVAYA CS1000 DIRECT SIP: IS THAT POSSIBLE? PLEASE HELP!

Hello,
I would like to ask really simple question here: is that possible to establish (configure, setup, create, make etc.) a direct SIP connection between LYNC 2010 and AVAYA CS1000 PBX. On one side I have Lync 2010 Standard Edition acting as Front-End Server
with co-located Mediation Server. On another side I have AVAYA CS1000 IP PBX Version 7.0. We are trying to make this happen for 1 full year with no luck. The issue is that we do not manage our PBX (external company does).
Here is what we are basically trying to achieve:
All I want is that basically LYNC user (phone # 5678) can dial PBX user (phone # 1234) and talk. Also PBX user can call LYNC user and they can talk. I'm pretty sure that all I'm asking is very simple thing. I don't expect magic to happen (e.g. when one user
calls another user the world will change). All I want is to be able to call between LYNC and PBX users so they can talk.
Sorry for repeating again but here is my simple question: can direct SIP (or call it SIP trunk if this is the right word) be established so LYNC and PBX users can call each other and talk. Whatever protocol is (G.711, G.722 etc.) I will be happy.
If you can let me know I would very much appreciate this. I'm sure I'm not alone in this world and many companies out there using AVAYA CS1000 and deployed LYNC and probably have the same question so it may be helpful to other people as well.
Thanks!

Hi Thamara for reply. I can see where you are going, but the challenge here is that I have found many threads including in this forums from people who successfully connected CS1000 PBX to Lync 2010 for two-way calling, and even dual-forking (which I cannot
even dream about at this time).
If you go to AVAYA WEB site, you can find official AVAYA compatibility MATRIX that is available here:
http://www.google.ca/url?sa=t&rct=j&q=avaya%20compatibility%20matrix%20lync&source=web&cd=1&ved=0CBwQFjAA&url=http%3A%2F%2Fsupport.avaya.com%2Fcss%2FP8%2Fdocuments%2F100135385&ei=v3-5TsXVKMLX0QGHpdSkCA&usg=AFQjCNG6cmW5izdOASMuW34b8iK0vWnwuw
According to this document, only CS1000 version 7.5 supports Lync 2010 and only via ACE - starting first quarter of 2011 (which is done deal by now) - see below for details. The problem is that I have a hard time to understand the meaning of the word "Supported".
I do not really need full support, all I want is to be able to call in both directions using 2 distinct numbers (e.g. 4 digits extensions) that are configured in both systems.
Another thing is this ACE. Apparently it is a big monster software that you need to buy and pay big bucks and lots of configuration just to make it working. But I'm not even sure what will work. We do have one-way calling (from Lync to PBX) - it just does
not work other way, and I have no idea why. I'm not sure if this cannot be working, or because it was not setup right. As I mentioned before we have no access and clue about IP PBX - this is one of my issue here. Withoul knowledge how it works on PBX side
I cannot be sure what is causing this.
So I was wondering if anybody in this world was able to connect CS1000 PBX to Lync and perform two-way calling using two systems.
Thanks!
Microsoft OCS 2010 Lync
Supported (via ACE) - 1Q11

CME 7.1 with SCCP 7940G phones and SIP connection to a VOIP provider - inbound outbound fails

Here's a quick and dirty diagram of a CME 7.1 configuration. The phone can all call each other but something is not quite right with the SIP provider. The registrar and SIP registration pieces are working but most of the configuration examples that I've seen make me think that the CME router was being used as the edge device to the internet. From my drawing, you can see that is not the case here. My edge device is a Cisco ASA5505 with 9.2.x software running. I might be missing something in the SIP gateway knowledge department. Without diving into the configuration, I'm wondering if SIP messages are failing for calls because of NAT'ing? Trying to do searches has been tricky because I keep running into information that is more about setting up CME for SIP phones or just getting SIP to work between CME and a SIP provider. I have that part working. I'm just a bit unsure about how an SCCP 7940G gets an outbound call or even gets one to come in.
When I dial from my cell phone to the pilot number, there are no rings, it just goes to the VOIP provider's voice mail. When I try to dial out, I get a fast busy.
So, is NAT a consideration? Will the SIP gateway set up a call (forward) via the pre-established SIP connection? Yeah, I do sound like a newb.
If anyone has good information about, let's say, an inbound call and how that traffic flow works.
Thanks!

Have you configured your ASA to either NAT the IP address of the CME router or to do port forwarding for port 5060?

Configure Jabber for Mac with local CUCM and WebEx Connect

Hi, I was wondering if anyone has been able to configure the Jabber for Mac 8.6.2 client to use the WebEx Connect presence server with a local CUCM and Unity Connection servers. The preferences accounts tab does not show or allow the addition of the voice services. I have added from a working local CUCM preferences plist file what I believe are the correct entries however I still cannot see the accounts on the preferences tab. We currently do not have a local cisco presence server hence the requirement to trial the WebEx Connect server as I can't get past the first configuration step without it.
regards
paul

Hi - I have done this via the admin portal but still cannot get the Jabber client for Mac to register for voice. The Windows version works fine for the same user and CUCM device. Are there any other settings that need to be enabled specific to the Mac client?
Thanks.

BE5K to Virtual CUCM and Unity Connection 10.5(2)?

Hi,
I am migrating a CUCM and Unity Connection 8.6 from physical server to CUCM and Unity 10.5(2) and I estimate about 2-hour effort with PCD to migrate the CUCM to be ready for testing. Not sure about Unity Connection Part though. The physical server is serving 120 users with 3 remote locations(via MGCP) and complex integration with other systems. I only done greenfield deployment before...no migration/upgrade experience...
Simple/not simple questions:
1. Will the 2-hour be enough for CUCM?
2. PCD doesnot seem support migration of Unity Connection to 10.5. What is the recommended approach? Gradually move data or cutover?
3. From your experience, how long it takes for upgrading/migrating Unity Connection?
I have these 2 documents from Cisco as my reference, any other recommendations?
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/ucmapMigrate/10_5_1/CUCM_BK_M24251C0_00_migrate-procedure-for-cucm_1051/CUCM_BK_M24251C0_00_migrate-procedure-for-cucm_1051_chapter_01.html
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/upgrade/guide/10xcucrugx/10xcucrug022.html

Thanks for pointing it out. I did not realize BE5K is not supported by PCD.
Will this be my next best bet then:
Install version 8.6.2 VM in ESXi
Export info using BAT from BE5K
Import info into 8.6.2 VM on ESXi
Upgrade the 8.6(2) VM to v10.5.x (via PCD I guess)
Test then discconnect BE5K
If I am correct, will 4-hour be enough?
I also saw some other suggested to do in-software upgrade to 9.x than export... I just feel that is too risky...BE5K is already a single point of failure what happens if in-software upgrade failed...

Telco Messages via PRI (VG) connected to CUCM 9.X via SIP Trunk??

Hello
Questions:
Should I hear tel-co message on an IP Phone if a call that is meant to be long distance is sent to the gateway without a one (1). Currently these calls simply ring until disconnect, but work properly if the user dials a 1, the user expects to get a message from the provider and I am wondering if the SIP trunk between GW and CUCM is not allowing it?
Scenario:
IP Phone --> CUCM (SIP Trunk) --> ISR 2901(PRI) --> PSTN
In CUCM we have a local pattern 9.[2-9]XX[2-9]XXXXXX
If the IP Phone dials a number that is actually a long distance number, which would require a 1, but they dial it as a 10 digit local number, the call gets to the gateway and IP Phone hears ringing but it, rings until it eventually disconnects. At this point I believe the gateway is sending a cause code back to CUCM and I would expect error message back from the telco informing the caller of the issue, such as "Please dial 1 before a long distance number". Is there a specific setting on the SIP trunk and or Gateway to achieve this?
See Q931 Debug
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9838F
Exclusive, Channel 15
Display i = 0xB1, 'London Hydro'
Calling Party Number
UCS5-GW-02# i = 0x2181, '5193334444'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '5191112222'
Plan:ISDN, Type:National
*Jul 23 15:33:28.363: ISDN Se0/0/1:23 Q931: RX <- STATUS pd = 8 callref = 0xC3FA
Cause i = 0x80AB28 - Access information discarded
Call State i = 0x01
*Jul 23 15:33:28.411: ISDN Se0/0/1:23 Q931: RX <- CALL_PROC pd = 8 callref = 0xC3FA
Channel ID i = 0xA9838F
Exclusive, Channel 15
*Jul 23 15:33:28.415: ISDN Se0/0/1:23 Q931: RX <- PROGRESS pd = 8 callref = 0xC3FA
Cause i = 0x80FF - Interworking error; unspecified
Progress Ind i = 0x8088 - In-band info or appropriate now available
Thanks
Richard

Yes, Early media needs to be turned on the SIP Profile that is associated to the SIP trunk. The setting is SIP Rel 1xx options. It needs to be set for Send PRACK for all 1XX messages. If you have CUCM 7.x , this setting is a service parameter.

6300i and VoIP/SIP connections

I' v just bought 6300i and try to configure connection to my VoIP/SIP provider and was surprised that a can only add providers listed by NOKIA.
Can anyone know how to add account to VoIP/SIP provider not listed on NOKIA web page (display when you try to add new account)?
Solved!
Go to Solution.

Yes there is a method to add any VoIP provider to 6300i. Please read this thread for more information,
http://discussion.forum.nokia.com/forum/showthread.php?t=136775

CUCI-Lync 9.2 with CUCM 8.6 - Cisco Unity Connection Visual Voicemail Not Working

Hi
I have CUCM and CUC 8.6.2 running and MOC with CUCI-Lync 8.5 (with visual Voicemail) running OK with full registry configuration (see below). We are moving to Lync 2013 and want to use CUCI-Lync 9.2.
A basic install of CUCI-Lync 9.2 works fine with CUCM (with manual setup of TFTP, CCMIP and CTI) but not with CUC. I can only call the VM Pilot but I don't get my visual voicemail. In the CUCI-Lync parameters I type in my CUC server IP adress and credentials but get a message saying that it can't connect.
As the config guide describes a config with CUCM 9 (with UC services) which I don't have in V8.6. I've tried using the old registry configuration or no registry configuration at all, I can't get CUCI-Lync to connect to CUC...
Has anyone done this ? Any suggestions ?
OLD REG Configuration:
Windows Registry Editor Version 5.00
[HKEY_CURRENT_USER\Software\Cisco Systems, Inc.\Unified Communications\CUCIMOC]
"RememberMe"=dword:00000001
"AutoLogin"=dword:00000001
[HKEY_CURRENT_USER\Software\Cisco Systems, Inc.\Client Services Framework\AdminData]
"TftpServer1"="1.1.1.1"
"TftpServer2"="2.2.2.2"
"TftpServer3"=""
"UseCUCMGroupForCti"="1"
"CcmcipServer1"="1.1.1.1"
"CcmcipServer2"="2.2.2.2"
"CcmcipServerValidation"="0"
"CsfStatsServer"=""
"CsfStatsCollectionEnabled"=""
"EnableNativeDirectoryProvider"="1"
"VoicemailPilotNumber"="12345"
"VoiceMailService_UseCredentialsFrom"="PHONE"
"VVM_SystemServer_0"="3.3.3.3"
"VVM_SystemServer_1"="4.4.4.4"
"VVM_SystemServer_VmwsProtocol_0"="HTTP"
"VVM_SystemServer_VmwsProtocol_1"="HTTP"
"VVM_SystemServer_VmwsPort_0"="80"
"VVM_SystemServer_VmwsPort_1"="80"
"VVM_Mailstore_Server_0"="3.3.3.3"
"VVM_Mailstore_Server_1"="4.4.4.4"
"VVM_Mailstore_ImapProtocol_0"=""
"VVM_Mailstore_ImapProtocol_1"=""
"VVM_Mailstore_ImapPort_0"="143"
"VVM_Mailstore_ImapPort_1"="143"
"VVM_Mailstore_InboxFolderName"=""
"VVM_Mailstore_EncryptedConnection"=""
"VVM_Mailstore_PollingInterval"=""
"AutomaticDeviceSelectionMode"="0"
"SSO_Enabled_CUCM"="false"
"DeviceProviderServer1"="1.1.1.1"
"DeviceProviderServer2"="2.2.2.2"
"DeviceProviderServerValidation"="0"
"DeviceProviderType"="CCMIP"

The UC Services are a CUCM 9.0 feature. In 8.x these existed within CUPS under Applications > CUPC/Jabber > CTI Gateway and Profile. Other things that frequently cause this to break: 1) deskphone not associated to your end user object; 2) primary extension not set; 3) standard cti enabled and standard ccm end users group membership missing; 4) the IP/FQDN of the CTI Gateway is not a CUCM node running CTI Manager.
Please remember to rate helpful responses and identify helpful or correct answers.

Doubt about integration CUCM and VoIP Provider

Hi Guys,
I have the follow doubt: Is possible to do integration CUCM and VoIP Provider using authentication? Is there necessary another equipment to do this? CUBE for example? or another alternative
Thanks,
Wilson

If you are referring to authentication over sip trunk, then NO! CUCM doesn't do authentication you will need a CUBE for that

CUCM and Jabber guest configuration

Hi Everybody ,
I would like to ask the cucm and jabber guest how configuration, I can't find the document to cisco.

Hi,
As per the link that i shared "In a production environment, Cisco Jabber Guest requires that your Cisco Unified Communications Manager be configured to work with Cisco Expressway.
Cisco Jabber Guest can be pointed directly to Cisco Unified Communications Manager for lab deployments only. You must configure a SIP trunk on Cisco Unified Communications Manager for this deployment. This option is best suited to a lab deployment in which the goal is to familiarize yourself with Cisco Jabber Guest without the additional overhead of configuring Expressway. However, without configuring Expressway, Cisco Jabber Guest is not supported in a production environment."
HTH
Manish

Siebel and Avaya IC 7.1 integration

Hi guys. Do you have some document that explain step by step the integration with siebel 7.x and avaya ic 7.x ?? Tks a lot.

CTI integration has its official documentation.
To sum it up:
General architecture "User oriented" /
-An Application Object Manager (AOM) display a Communication toolbar (a Tools "customizable" java applet)
-A user having a Communication profile will see this toolbar
-Clicking on the toolbar will usually just send a CTI event to the CTI server
-Sending an event means starting a short CommSessionMgr task
-The CommSessionMgr is where the Avaya DLL/EXE got executed
-It's able to RECEIVE CTI events from the server and dispatch those to the AOM
Installing the Avaya drivers should be ok /
Parameterization of the connection between SIEBEL and the CTI server stands at the Configuration /
Customization of the CTI behaviour means Tools for the toolbar and the WebClient to manage (a .DEF file allows to delivers these configurations) how SIEBEL reacts to CTI events
You need specific experts for:
-CTI architecture (even alone it may hurts)
-SIEBEL to CTI server driver's parameter sizing and validation (very rare...I am one ;-)
-SIEBEL Configuration of the events (Oracle got some and some consulting or service companies as well)

Cisco CUCM to Alcatel PBX SIP calling issues

Hi All
I have configured a SIP trunk between my cucms and an Alcatel old pbx on a remote site, they are all identical configs.
However one of them, the remote site Alcatel can call my cucm and voice is ok
But when we try to dial from the CUCM to the Alcatel we are getting the fast busy tone!
Codecs are set etc! as it works one way fine!
any ideas what thsi could be ?
cheers

Hi here is a snip of the trace for the call
the calling phone was ext 448 the called number over the sip trunk is 88044615
cheers
16
2015/01/23 08:15:31.897|CC|REJECT|26821723|26821724|476|8804615|8804615|1
2015/01/23 08:15:43.577|CC|RELEASE|26821726|26821727|16
2015/01/23 08:15:54.907|CC|SETUP|26821728|26821729|476|88044615|88044615
2015/01/23 08:15:54.909|CC|OFFERED|26821728|26821729|476|88044615|88044615|SEPC4641301122E|DELHI-SIP-TRUNK
2015/01/23 08:15:55.273|SIPT|26821729|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,13,53766610.2^*^*|13409968|[email protected]|INVITE
2015/01/23 08:15:55.640|SIPT|26821729|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651678^172.20.65.5^*|13409969|[email protected]|100 Trying
2015/01/23 08:15:56.012|SIPT|26821729|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651679^172.20.65.5^*|13409970|[email protected]|403 Forbidden
2015/01/23 08:15:56.012|SIPT|26821729|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651679^172.20.65.5^*|13409971|[email protected]|ACK
2015/01/23 08:15:58.668|CC|RELEASE|26821728|26821729|67108885
2015/01/23 08:16:02.133|CC|SETUP|26821730|26821731|4133320804|467|467
2015/01/23 08:16:02.136|CC|OFFERED|26821730|26821731|4133320804|467|467|172.24.32.38|SEPC464130114C0
2015/01/23 08:16:18.712|CC|SETUP|26821733|26821734|568|487|487
2015/01/23 08:16:18.714|CC|OFFERED|26821733|26821734|568|487|487|SEPC464130117E9|SEPC4641301147E
2015/01/23 08:16:20.151|CC|SETUP|26821730|26821737|4133320804|467|1999
2015/01/23 08:16:20.157|CC|OFFERED|26821730|26821737|4133320804|467|1999|172.24.32.38|CiscoUM1-VI54
2015/01/23 08:16:20.159|CC|RELEASE|26821731|0|0
2015/01/23 08:16:28.151|CC|RELEASE|26821730|26821737|16
2015/01/23 08:16:31.997|CC|SETUP|26821738|26821739|476|88044615|88044615
2015/01/23 08:16:31.998|CC|OFFERED|26821738|26821739|476|88044615|88044615|SEPC4641301122E|DELHI-SIP-TRUNK
2015/01/23 08:16:31.999|SIPT|26821739|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,13,51956482.95772^172.24.48.180^SEPC4641301122E|13409978|[email protected]|INVITE
2015/01/23 08:16:32.366|SIPT|26821739|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651682^172.20.65.5^*|13409979|[email protected]|100 Trying
2015/01/23 08:16:32.764|SIPT|26821739|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651683^172.20.65.5^*|13409980|[email protected]|403 Forbidden
2015/01/23 08:16:32.764|SIPT|26821739|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651683^172.20.65.5^*|13409981|[email protected]|ACK
2015/01/23 08:16:39.148|CC|RELEASE|26821738|26821739|67108885
2015/01/23 08:16:44.261|CC|SETUP|26821740|26821741|4133320804|465|465
2015/01/23 08:16:44.263|CC|OFFERED|26821740|26821741|4133320804|465|465|172.24.32.38|SEPC46413011466
2015/01/23 08:17:26.622|CC|RELEASE|26821733|26821734|16
2015/01/23 08:17:33.320|CC|SETUP|26821743|26821744|568|536|536
2015/01/23 08:17:33.322|CC|OFFERED|26821743|26821744|568|536|536|SEPC464130117E9|SEPC46413011477
2015/01/23 08:17:44.673|CC|RELEASE|26821706|26821707|16
2015/01/23 08:18:38.248|CC|RELEASE|26821713|26821714|16
2015/01/23 08:18:51.306|CC|RELEASE|26821740|26821741|16
2015/01/23 08:18:53.509|CC|SETUP|26821746|26821747|447|033255961|033255961
2015/01/23 08:18:53.513|CC|OFFERED|26821746|26821747|447|033255961|033255961|SEPC46413011482|172.24.32.38
2015/01/23 08:18:53.739|SIPL|0|TCP|IN|172.24.32.34|50

RE: POTS-to-any and any-to-POTS connections are enabled

Hi,
First post here. Thanks for the assistance.
On
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0in 5.4pt 0in 5.4pt;
mso-para-margin:0in;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:"Times New Roman";
mso-fareast-theme-font:minor-fareast;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;}
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-h323h323.html , it indicates this comment
"Note H.323-to-H.323: By default, H.323-to-H.323 connections are disabled and POTS-to-any and any-to-POTS connections are enabled"
My query is what is POTS? On the page itself, there is no definition. I try keying in POTS into the search bar but still no answers.
Anyone knows?

Hi Roman,
Thanks for the quick reply
The reason I'm asking is my vendor is recommending that we include these into our router
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0in 5.4pt 0in 5.4pt;
mso-para-margin:0in;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:"Times New Roman";
mso-fareast-theme-font:minor-fareast;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;}
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
But do I have to since it indicates
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0in 5.4pt 0in 5.4pt;
mso-para-margin:0in;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:"Times New Roman";
mso-fareast-theme-font:minor-fareast;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;}
By default, POTS-to-any and any-to-POTS connections are enabled.

Voip between cisco and avaya

We have to implement Voice over IP on our IPLC circuit between foreign office and India office.
Topology is such that our foreign office is having 1700 series router with wan card
(Wic-1T) and one two port FXO card connected to PBX for PSTN and local connectivity.
India end we have a cisco 1700 router with wan card (Wic-1T) and on lan we have Avaya S8300 media server with multiple phones connected to it. Avaya server is acting as a voice gateway at India end. Avaya server is connected to LAN on Ethernet.
We have configured IPSEC site-to-site VPN between both the locations.
Requirement:-
Over the VPN tunnel we want to enable voice communication between both the locations.
i.e between cisco at foreign end and avaya server at India end.
1) India office users via IPLC should communicate over voice and data with the foreign office (Avaya extensions to FXO in Foreign office).
2) Foreign office should communicate with the india office users connected to IPLC link (FXO to Avaya server extensions)
Any suggestions/config will be appreciated
Attaching the india end router config.
Thanks

Hi
Before getting onto the requirements,from now on i would suggest you to mask the ip (public) address part included in the configs while posting in any general forum.
These are the links which you can use up here for the ref..
You need to check out for the possible codes and other compatible points which you need to configure on both the sidees..
http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a008009431b.shtml
http://www.cisco.com/en/US/products/hw/routers/ps259/products_configuration_guide_chapter09186a008007e606.html
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_configuration_guide_chapter09186a0080080afd.html
i m not sure though about the configuration of Avaya box but you can mention the remote end routers ethernet ip as the destination ip or the target for the call termination.
And in the remote router while configuring the dial-peer pots and dial-peer voip binded on the FXO itnerface make sure that you are mentioning the Avaya box ip as the session target.
regds

CUCM and Avaya CS1000 SIP connection

Similar Messages

Maybe you are looking for